similar to: SIP phone behind NAT

Hi all, I've just been setting up such a phone with * and I'm facing some problems. At this time there is no customized extensions & others, so I use extension 8500 (voicemail) in default config. What I get when this extension is called from the SNOM is pasted below. The phone ignores *'s invites and * sends a "BYE" to it. Please note the IP in the "Contact"

I'm trying to get a Snom100 configured with H.323. Right now, the phone is not even connecting to the Asterisk server, so there's obviously a problem with the snom config. Does anybody have a sample working configuration with the snom phone, using H.323? I've checked the archives, but everybody seems to be using SIP with the Snom phone, not H.323. -Tilghman

Hi all, ------ I tried to use * over satellite, but all my effort did not succeed. The Asterisk is behind the VSAT and is resposibel for alle the SIP clients in a field location. The clients are notebooks and PDA's running SJPhoen for Windows and PocketPC. Unfortunately I could not find any Linux Client wich worked satisfying. SJ LAbs promised a Linux Version at the end of August but they

sorry, this problem is fixed by myself. we must need set 'canreinvite=no' each user. --- I'm try to discconect a call with SIP. when caller make a call, 'show channels' result is following. mack*CLI> show channels Channel (Context Extension Pri ) State Appl. Data SIP/mack-1bfc (default 1 ) Ringing AppDial (Outgoing

OK I did some researches and tests with it, and finally: I registered my messenger to the asterisk and called if from the snom. The audio from the snom to the messenger was PERFECT. By the time of the call This message was running on the asterisk console: WARNING[16400]: File dsp.c, Line 1107 (ast_dsp_process): Unable to detect process 2 frames My conclusion is that the snom100 utilizes MSGSM

Hi, there were some postings a few weeks ago telling that the GSM codec problem with snom100 will be fixed. But it still seems to be very quality. Will be any change in this subject? THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/2fa9d206/attachment.htm

hello, can anybody help me with folloving problem I have asterisk with the public IP and two UAs (snom100, x-lite) in the same private network behind the same FW/NAT. All is working good, but whan I tried to establish call between these two UAs, first 10-15 second is nothing to hear and then is the quality terrible :( Can anyone tell how to get it work with normal quality ? best regards

Hi all, I have been struggling desperately to get * work together with my snom100 for days on end, but I am not making any progress... Of the entries marked *#) I'm still not sure what it does; so far I have on the snom in "SIP/lines" -user name - empty *1) -account - Conrad -registrar - 192.168.200.83 -action - "None" *2) in

Hi folks, in order to establish backward compatibility we made an image that automatically detects if the other side does not support RFC3264. Please try it out, we would be very interested if this image is a progress! http://snom.com/download/share/snom200-2.03b-SIP.bin Thanks, CS

Hello. I've been going through the archives, but can't discern the state or future direction of IAX on the SNOM100. The most recent image appears to be from September 2002. There was a message on this list stating that SNOM was coming to visit Digium last April with the intention of adding IAX support themselves. For a while there was reference to the I100E on the asterisk and/or

Hi, Just wondering if anyone else is able to reproduce this with the current * (CVS 12:15 GMT) Call Snom from any device (tested with i4l, zaptel and SIP). Answer call and try to transfer call using transfer button on Snom. After dialing new number press OK (F4). At this moment the Snom users hears dialtone, but the caller still hears the Snom user... Even hangup on the Snom doesn't

Hello. I have two snom200 ip phones and 1 mp108fxs (audiocodes 8 fxs) and i dont get a ring in the caller phone when I dial from a snom200 to the other snom200 or the mp108fxs, I made a debug with ethereal, and I can see a "Ringing" packet being return from the called snom200 or mp108fxs to the asterisk box, but it is not being re-transmitted to the caller snom200. Altough

hi, can some one tell me a good IP phone (not software, but a "real" phone :) that work well with asterisk? how mutch does it cost a good IP phone? i made a VoIP network for my company, but now we are using a client for PC phone... i'd like to buy a IP phone, can someone tell me witch model i should buy? thanks, Angelo

I am using the snom 200 with Phone type snom200-SIP Version snom200-SIP 2.04g Bootloader URL http://www.snom.com/download/snom200-boot1.9.bin Firmware URL http://www.snom.com/download/share/snom200-2.04o-SIP.bin I am using asterisk stable tree. I had to disable "Challenge Response on Phone" on my snom; I could not get it to work with

Check the Redirection on the web interface of your Snom 200. If it says "When Busy" that's your problem. It should say "Never". Also make sure on Sip->Lines your line appearance says "All" or else you will have the same problem. Hope this helps :) Greg Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original

The snom 200 and 220 have five programmable buttons. Each button has a led that can be used to indecate if an extension is idle, in use, or ringing. A button pannel for the 220 is also comming out soon that will have 20'ish programmable buttons on board. This is a killer app for any company that has receptionists handle calls, and pretty usefull for everyone else. As a matter of fact,

Hello list, i am using: asterisk CVS-10/13/03-11:54:33 chan_capi-0.3.0 ATA-186 V2.16.1.ms over MGCP Situation: ISDN calls ATA ISDN speaks with ATA ATA-Phone presses Flash and speaks to another one (SIP/snom200) ATA-Phone hangs up ISDN talks to SIP/snom200 snom200 hangs up The incoming extension of ATA keeps busy for a time (20 sec?), even its not off-hook anymore! Any ideas? -- Swapping

Try this one. Took me a while too. http://www.snom.com/download/share/snom200-2.05c-SIP.bin > -----Original Message----- > From: M3 Freak [mailto:m3freak@rogers.com] > Sent: Monday, May 17, 2004 11:42 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Snom200 Firmware: I only see 2.04g > > Hello all, > > I've noticed several messages about the

Hi Steve, I am having this problem in which RxFax is still holding the file after receiving a complete fax. Somehow the zap channel is still active but on the fax client it was sent successfully. If you call the line it is still busy. Changed from phase 3 to 4 >>> MCF: 8c HDLC underflow in state 8 Changed from phase 4 to 3 Slow carrier up <<< DCN: fb DCN with final frame tag

hi all when starting kphone, it tries to register with asterisk but fails after a while. The SIP entry in * for this user is below. This is identical to the other SIP entries. The other SIP clients are MSN messenger plus one snom. these work fine. See SIP debug output attached as 'screen-exchange' thanks roy [roy] type=friend ;insecure=yes username=roy ;secret=password host=dynamic